- Runs in the browser and Node.js.
- SIP over WebSocket (use real SIP in your web apps)
- Audio/video calls (WebRTC) and instant messaging
- Lightweight!
- Easy to use and powerful user API
- Works with OverSIP, Kamailio, Asterisk. Mobicents and repro (reSIProcate) servers (more info)
- Written by the authors of RFC 7118 "The WebSocket Protocol as a Transport for SIP" and OverSIP
Starting from 3.0.0, JsSIP no longer includes the rtcninja module. However, the jssip-rtcninja package is based on the 2.0.x branch, which does include rtcninja.
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For questions or usage problems please use the jssip public Google Group.
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For bug reports or feature requests open an Github issue.
The following simple JavaScript code creates a JsSIP User Agent instance and makes a SIP call:
// Create our JsSIP instance and run it:
var socket = new JsSIP.WebSocketInterface('wss:https://sip.myhost.com');
var configuration = {
sockets : [ socket ],
uri : 'sip:[email protected]',
password : 'superpassword'
};
var ua = new JsSIP.UA(configuration);
ua.start();
// Register callbacks to desired call events
var eventHandlers = {
'progress': function(e) {
console.log('call is in progress');
},
'failed': function(e) {
console.log('call failed with cause: '+ e.data.cause);
},
'ended': function(e) {
console.log('call ended with cause: '+ e.data.cause);
},
'confirmed': function(e) {
console.log('call confirmed');
}
};
var options = {
'eventHandlers' : eventHandlers,
'mediaConstraints' : { 'audio': true, 'video': true